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Fanvil Product User Manual IP-Gateway Model: A1 Version:V1.0.60.16 © 2005 Fanvil technology Co., Ltd All rights reserved. This document is supplied by Fanvil Technology Co., Ltd, No part of this document may be reproduced, republished or retransmitted in any form or by any means whatsoever, whether electronically or mechanically, including, but not limited to, by way of photocopying, recording, information recording or through retrieval systems, without the express written permission of Fanvil Technology Co., Ltd. Fanvil Technology Co., Ltd reserves the right to revise this document and make changes at any time and without the obligation to notify any person and/or entity of such revisions and/or changes. Product specifications contained in this document are subject to change without notice. Safety Notices Please read the following safety notices before installing or using this gateway. They are crucial for the safe and reliable operation of the device. z Please use the external power supply that is included in the package. Other powers supplies may cause damage to the device, affect the behavior or induce noise. z Before using the external power supply in the package, please check with home power voltage. Inaccurate power voltage may cause fire and damage. z Please do not damage the power cord. If power cord or plug is impaired, do not use it, it may cause fire or electric shock. z The plug-socket combination must be accessible at all times because it serves as the main disconnecting device. z Do not drop, knock or shake it. Rough handling can break internal circuit boards. z Do not install the device in places where there is direct sunlight. Also do not put the device on carpets or cushions. It may cause fire or breakdown. z Avoid exposure the gateway to high temperature, below 0℃ or high humidity. z z z z z Avoid wetting the unit with any liquid. Do not attempt to open it. Non-expert handling of the device could damage it. Consult your authorized dealer for help, or else it may cause fire, electric shock and breakdown. Do not use harsh chemicals, cleaning solvents, or strong detergents to clean it. Wipe it with a soft cloth that has been slightly dampened in a mild soap and water solution. When lightning, do not touch power plug or device line, it may cause an electric shock. Do not install this device in an ill-ventilated place. You are in a situation that could cause bodily injury. Before you work on any equipment, be aware of the hazards involved with electrical circuitry and be familiar with standard practices for preventing accidents. Table of Content 1. WELCOME TO THE A1 SINGLE PORT GATEWAY.........................................................................................5 1.1. Package Contents....................................................................................................................5 2 UNDERSTANDING OF A1 SINGLE PORT GATEWAY ......................................................................................6 2.1. The positive of A1 single port gateway .................................................................................6 2.2. Indicator signs .........................................................................................................................7 2.3. Connector description ............................................................................................................7 3. GETTING STARTED ...............................................................................................................................9 3.1. Connect the power and network ............................................................................................9 3.1.1. Connect the network ........................................................................................................9 3.1.2. Connect the power ......................................................................................................... 10 4. A1 BASIC OPERATION OF A SINGLE PORT GATEWAY PHONE..................................................................... 11 4.1. Call transfer............................................................................................................................ 11 4.2. Call hold ................................................................................................................................. 11 4.3. With the PSTN user calls ...................................................................................................... 12 5. WEB CONFIGURATION ......................................................................................................................... 13 5.1. Introduction of configuration................................................................................................ 13 5.1.1. Ways to configure .......................................................................................................... 13 5.1.2. Password Configuration ................................................................................................ 13 5.2. Setting via web browser........................................................................................................ 13 5.3. Configuration via WEB.......................................................................................................... 14 5.3.1. BASIC .............................................................................................................................. 14 5.3.1.1. Status ........................................................................................................................... 14 5.3.1.2. Wizard........................................................................................................................... 15 5.3.2. Network ........................................................................................................................... 17 5.3.2.1. WAN Config.................................................................................................................. 17 5.3.2.2. LAN Config................................................................................................................... 19 5.3.2.3. Qos Config ................................................................................................................... 19 5.3.2.4. Service Port.................................................................................................................. 21 5.3.2.5. DHCP SERVER............................................................................................................. 23 5.3.2.6. NTP ...............................................................................................................................24 5.3.3. VOIP................................................................................................................................. 24 5.3.3.1. SIP Config .................................................................................................................... 24 5.3.3.2. Stun Config .................................................................................................................. 28 5.3.3.3. DIAL PEER setting....................................................................................................... 29 5.3.4. Phone .............................................................................................................................. 33 5.3.4.1. DSP Config................................................................................................................... 33 5.3.4.2. Call Service .................................................................................................................. 34 5.3.4.3. Digital Map Configuration ........................................................................................... 35 5.3.5. Maintenance.................................................................................................................... 37 5.3.5.1. Auto Provision ............................................................................................................. 37 5.3.5.2. Syslog Config .............................................................................................................. 38 5.3.5.3. Config Setting .............................................................................................................. 39 5.3.5.4. Update .......................................................................................................................... 39 5.3.5.5. Account Config............................................................................................................ 40 5.3.5.6. Reboot .......................................................................................................................... 41 5.3.6. Security ........................................................................................................................... 42 5.3.6.1. MMI Filter...................................................................................................................... 42 5.3.6.2. Firewall ......................................................................................................................... 43 5.3.6.3. NAT Config................................................................................................................... 44 5.3.6.4. VPN Config................................................................................................................... 46 5.3.7. Logout ............................................................................................................................. 47 6. APPENDIX ......................................................................................................................................... 48 6.1. SPECIFICATION ...............................................................................................................................48 6.1.1. HARDWARE ................................................................................................................................. 48 6.1.2. VOICE FEATURES ......................................................................................................................... 48 6.1.3. NETWORK FEATURES ................................................................................................................... 48 6.1.4. MAINTENANCE AND MANAGEMENT ................................................................................................. 49 6.2. PARTICULARLY SUITABLE FOR A1 SINGLE PORT GATEWAY ................................................................... 49 6.3. COMMON PROBLEMS ...................................................................................................................... 49 1. Welcome to the A1 single port gateway 1.1. Package Contents Please check your product packaging,it includes: 1. One A1 single port gateway 2. A group of cable 3. A power adapter NOTE: if you use the non-A1 single port gateway comes with a power adapter, single port gateway A1 may cause damage or other injury. Specifications for the power adapter may difference between the different ship areas,If the power adapter provided with the product can not be used locally, please consult your local dealer. 4. User manua 2 Understanding of a single port gateway A1 A1 Single-port gateway IP-based voice media access device is designed for operators, enterprises, residential users, and residential VoIP solution to provide network equipment. A1 single port gateway into the analog voice information transmitted over IP networks, which use IP networks to transmit voice. It is full compliance with the SIP protocol standard, with the market most other SIP compliant devices and server-side. The gateway will play Internet network (either public network or private network) connecting with the public telephone network bridge. It provides an FXS analog voice interface, used for ordinary small business PBX or gateway (PBX). Also provide an additional interface to a public telephone network PSTN (IE escape interface); power for the gateway, the call line will automatically go to PSTN lines from the VoIP line, the normal traffic for the user to provide the most effective protection. Off this site using the most advanced voice processing technologies, such as advanced voice compression standards, echo cancellation, dynamic voice detection, silence detection, ensuring Quality of Service (QoS), voice quality comparable to regular PSTN phone. In addition, A1 single port gateway also integrates a small router function. WEB comes through the gateway configuration page, simply configure the network parameters, can achieve multiple computers and network equipment, broadband access, ideal for small office and home users. Because this site has a wealth of features and related detailed configuration options, in your call to enjoy a stress free before you know your A1 single port gateway. 2.1. The positive of A1 single port gateway 2.2. Indicator signs Name POWER REG PHONE Meaning Power LED Registration status indicator Phone work status lights WAN WAN network interface lights LAN LAN network interface lights Description Always light, has power, you can start using the A1 single port gateway. Registered, the lights lit, the registration fails, the light flashes; do not use Notes Books, lights out. Show VoIP service is being used, or PSTN services, service, hang up: Death; pick up after the state if it is VoIP, Always; if you are in PSTN state: off. Indicator light, WAN port connected to the network. Flashing: Data transfer. Indicator light, LAN port connected to the network. Flashing: Data transfer. 2.3. Connector description Name POWER LINE PHONE Meaning Power switch Lifeline FXS Interface LAN Network Interface Network Interface WAN Description Output:12VDC,500mA。 PSTN access lines. Ordinary telephone connection, or switch into the line. 10/100M Adaptive connected PC. 10/100M Adaptive connected to the RJ45 port of Internet。 A1 single port gateway with two network interface itself: WAN port and LAN port, you can use the Internet connection into the WAN port or LAN port. Before inserting the power to read the manual carefully of “Safety." 3. Getting Started Before you start using the A1 single port gateway, please install the following: 3.1. Connect the power and network 3.1.1. Connect the network During this step, make sure your environment already have broadband Internet access capability. 1. Broadband Router Direct network connection—by this method, you need at least one available Ethernet port in your workspace. Use the Ethernet cable in the package to connect WAN port on the back of your phone to the Ethernet port in your workspace. Since this VoIP Phone has router functionality, whether you have a broadband router or not, you can make direct network connect. The following two figures are for your reference. 2. as a broadband router Use this method if you have a single Ethernet port in your workspace with your desktop computer already connected to it. First, disconnect the Ethernet cable from the computer and attach it to the WAN port on the back of your phone. Next, use the Ethernet cable in the package to connect LAN port on the back of your phone to your desktop computer. Your IP Phone now shares a network connection with your computer. The following figure is for your reference. 3.1.2. Connect the power During this step, make sure your power supply connector and a single port gateway A1 outlet match, while A1 is also in line voltage and current required for a single port gateway. 1. The transformer connected to the DC port on the back of A1 single port gateway POWER jack 2. The AC adapter plug to an electrical outlet, A1 single port gateway boot. 3. At this point all of your lights (except the POWER indicator) will flash together. After booting, you will hear popping sounds, and then the indicator light is lit according to your current configuration corresponding light. (If your light is not normal, you need to further configure your network connection mode) 4. If you login on the gateway server, then you can start calling 4. A1 Basic operation of a single port gateway phone 4.1. Call transfer z Blind Transfer During a call, press FLASH (Flash) key, enter the number to be transferred * add and press 【#】 key to confirm, you can transfer the current call to third parties. (To use this feature, you must enable the gateway of the Call Waiting and Call Transfer function) z Attended Transfer During a call, press FLASH (Flash) key, enter the number waiting to be transferred connected, directly hang up, you can transfer successfully. (To use this feature, you must enable the gateway of the Call Waiting and Call Transfer function) NOTE:1, Call Transfer must call in two cases all the way is free for operation; 2, Gateway (transfer side) and the establishment of phone A calls phone C gateway and then create a call, hang up the phone A, this time the gateway can also initiate the transfer. 3, your VoIP traffic services providers need to support (RFC3515), this feature to work correctly. 4.2. Call hold z Call Hold and set aside During a call you can press FLASH (FLASH) button and enter the number to dial and press 【#】 key to ensure Recognition, can retain the current state of the call with third-party calls. If you press the FLASH (Flash) key, you can switch back. You also can send and receive on one side, then the party can not be retained to hear your conversation, the speaker you can not. During a call if you press 【*】 operation, will enter the three-way calling mode. (To use this feature, you must enable the gateway of the Call Waiting feature, you must achieve three-way calling mode to start the gateway Three Way Call function) z Call on hold and accept call waiting In normal conversation, a third party dial-in, the handset will beep ~ beep ~ tips coming, you can use FLASH (Flash) button to accept call waiting. If you press this button again, you can switch back. You also can send and receive on one side, then the party can not be retained to hear your conversation, the speaker you can not. (To use this feature, you must enable the gateway of the Call Waiting feature) 4.3. With the PSTN user calls * T mapping shows that when the user connected to the PSTN line to the LINE port, then press * to switch to PSTN line, the user can call through the PSTN; if re-hook-hook dialing, the default line, or VoIP, need to press * to switch. Of course you can also set the others, does not necessarily use the * T (Finally, the T end) (See specific operations 5.3.3 Dial-peer) Lifeline of the main functions is: to prevent blackouts, No Network Under such circumstances, the availability of telephone remains! Now, when introduced Notes on using the lifeline. In two cases: z Gateway is taking the lifeline, the user can use it as a regular phone z Normal operating conditions, preferably up at the gateway before the PSTN line will be inserted into the escape port (LINE). If you plug in the PSTN line up after the gateway, then you may hear pops or two, then you can not busy with a lifeline, but should wait a few seconds, pops, etc. to hear the same twice. At this point, you can rest assured that use. 5. Web configuration 5.1. Introduction of configuration 5.1.1. Ways to configure A1 gateway has three different ways to different users. z Use phone keypad. z Use web browser(recommendatory way). z Use telnet with CLI command. 5.1.2. Password Configuration There are two levels to access to phone: root level and general level. User with root level can browse and set all configuration parameters, while user with general level can set all configuration parameters except SIP (1-2) or IAX2’s that some parameters can not be changed, such as server address and port. User will has different access level with different username and password. z Default user with general level: username:guest password:guest z Default user with root level: username:admin password:admin The default password of phone screen menu is 123. 5.2. Setting via web browser When this phone and PC are connected to network, enter the IP address of the wan port in this phone as the URL (e.g. http://xxx.xxx.xxx.xxx/ or http://xxx.xxx.xxx.xxx:xxxx/). Gateway IP address can be key by dialing # * 111 inquiries received The login page is as below picture NOTE: After you configure the gateway, you need click save button in config under Maintenance in the left catalog to save your configuration. Otherwise the device will lose your modification after power off and on. 5.3. Configuration via WEB 5.3.1. BASIC 5.3.1.1. Status Status Field name Network Explanation Shows the configuration information on WAN and LAN port, including the connect mode of WAN port (Static, DHCP, PPPoE), MAC address, the IP address of WAN port and LAN port, ON or OFF of DHCP mode of LAN port. Phone Number Shows the phone numbers provided by the SIP LINE 1-3 servers and IAX2. The last line shows the version number and issued date. 5.3.1.2. Wizard Wizard Field Name Explanation Please select the proper network mode according to the network condition. A1 gateway provide three different network settings: z Static: If your ISP server provides you the static IP address, please select this mode, and then finish Static Mode setting. If you don’t know about parameters of Static Mode setting, please ask your ISP for them. z DHCP: In this mode, you will get the information from the DHCP server automatically; need not to input this information artificially. z PPPoE: In this mode, your must input your ADSL account and password. You can also refer to 3.2.1 Network setting to speed setting your network. Choose Static IP MODE,click【NEXT】can config the network and SIP(default SIP1)simply, also can browse too. Click【BACK】can return to the last page. Static IP Address Netmask Gateway DNS Domain Primary DNS Alter DNS Input the IP address distributed to you. Input the Netmask distributed to you. Input the Gateway address distributed to you. Set DNS domain postfix. When the domain which you input can not be parsed, gateway will automatically add this domain to the end of the domain which you input before and parse it again. Input your primary DNS server address. Input your standby DNS server address. Display Name Server Address Server Port User Name Password Phone Number Enable Register Set the display name. Input your SIP server address. Set your SIP server port. Input your SIP register account name. Input your SIP register password. Input the phone number assigned by your VOIP service provider. Start to register or not by selecting it or not. Display detailed information that you manual config. Choose DHCP MODE,click【NEXT】can config SIP(default SIP1)simply, also can browse too. Click【BACK】can return to the last page. Like Static IP MODE。 Choose PPPoE MODE,click【NEXT】can config the PPPoE account/password and SIP(default SIP1)simply, also can browse too. Click【BACK】can return to the last page. Like Static IP MODE。 PPPoE Server It will be provided by ISP. Username Input your ADSL account. Password Input your ADSL password. Notice: Click【Finish】button after finished your setting, gateway will save the setting automatically and reboot, After reboot, you can dial by the SIP account. 5.3.2. Network 5.3.2.1. WAN Config WAN Config Field Name Active IP Current Netmask MAC Address Current Gateway Get MAC Time explanation The current IP address of the gateway. The current Netmask address. The current MAC address of the gateway. The current Gateway IP address. Shows the time of getting MAC address Please select the proper network mode according to the network condition. A1 gateway provide three different network settings: z Static: If your ISP server provides you the static IP address, please select this mode, and then finish Static Mode setting. If you don’t know about parameters of Static Mode setting, please ask your ISP for them. z DHCP: In this mode, you will get the information from the DHCP server automatically; need not to input this information artificially. z PPPoE: In this mode, your must input your ADSL account and password. You can also refer to 3.2.1 Network setting to speed setting your network. Obtain DNS Select it to use DHCP mode to get DNS address, if you server don’t select it, you will use static DNS server. The automatically default is selecting it. If you use static mode, you need set it. IP Address Input the IP address distributed to you. Netmask Input the Netmask distributed to you. Gateway Input the Gateway address distributed to you. Set DNS domain postfix. When the domain which you DNS Domain input can not be parsed, gateway will automatically add this domain to the end of the domain which you input before and parse it again. Primary DNS Input your primary DNS server address. Alter DNS Input your standby DNS server address. If you uses PPPoE mode, you need to make the above setting. PPPoE Server It will be provided by ISP. Username Input your ADSL account. Password Input your ADSL password. Notice: 1)Click “Apply” button after finished your setting, IP gateway will save the setting automatically and new setting will take effect. 2)If you modify the IP address, the web will not response by the old IP address. Your need input new IP address in the address column to logon in the web. 3)If networks ID which is DHCP server distributed is same as network ID which is used by LAN of system, system will use the DHCP IP to set WAN, and modify LAN’s networks ID(for example, system will change LAN IP from 192.168.10.1 to 192.168.11.1) when system uses DHCP client to get IP in startup; if system uses DHCP client to get IP in running status and network ID is also same as LAN’s, system will refuse to accept the IP to configure WAN. So WAN’s active IP will be 0.0.0.0 5.3.2.2. LAN Config LAN Config Field name LAN IP Netmask DHCP Service NAT Bridge Mode explanation Specify LAN static IP. Specify LAN Netmask. Select the DHCP server of LAN port or not. After you modify the LAN IP address, gateway will amend and adjust the DHCP Lease Table and save the result amended automatically according to the IP address and Netmask. You need restart the gateway and the DHCP server setting will take effect. Select NAT or not. Select Bridge Mode or not: If you select Bridge Mode, the gateway will no longer set IP address for LAN physical port,LAN and WAN will join in the same network. Click “Apply”, the gateway will reboot. Notice: If you choose the bridge mode, the LAN configuration will be disabled. 5.3.2.3. Qos Config The gateway support 802.1Q/P protocol and DiffServ configuration. VLAN functionality can use different VLAN IDs by setting signal/voice VLAN and data VLAN. The VLAN application of this device is very flexible. In chart 1, there is a layer 2 switch without setting VLAN. Any broadcast frame will be transmitted to the other ports except the send port. For example, a broadcast information is sent out from port 1 then transmitted to port 2,3and 4. In chart 2, red and blue indicate two different VLANs in the switch, and port 1 and port 2 belong to red VLAN, port 3 and port 4 belong to blue VLAN. If a broadcast frame is sent out from port 1, switch will transmit it to port 2, the other port in the red VLAN and not transmit it to port3 and port 4 in blue VLAN. By this means, VLAN divide the broadcast domain via restricting the range of broadcast frame transmit ion. Note: chart 2 use red and blue to identify the different VLAN, but in practice, VLAN uses different VLAN IDs to identify. QoS Configuration Field name VLAN Enable VLAN ID Enable Check Voice/Data VLAN differentiated explanation Before select it to enable VLAN, you need enable Bridge mode in LAN config. Enable VLAN ID check by selecting it. After enable VLAN ID check, if VLAN ID of a data package is not the same with the gateway or a data package do not have VLAN ID, the data package will be discarded. After enable VLAN, system will set packets with different type of VLAN ID. Undifferentiated means after using VLAN, both VoIP packets and other data packets will use the voice VLAN ID; tag differentiated means after using VLAN, VoIP(signal and voice) packets will DiffServ Enable DiffServ Value Voice 802.1P Priority Data 802.1P Priority Voice VLAN ID Data VLAN ID add voice VLAN ID, and other data packets will add data VLAN ID; data untagged means after using VLAN, only VoIP packets will add voice VLAN ID. Other data packets will not use VLAN. Select it or not to Enable or disable DiffServ. Set DiffServ value, the common value is 0x00. Specify 802.1P Priority of voice/signal data package. Set 802.1p of data VLAN. Non-VoIP data (such as http, telnet, ping etc) will use this value to set VLAN package. Set VLAN ID of voice/signal data package. Set 802.1q of data VLAN ID. Non-VoIP data (such as http, telnet, ping etc) will use this value to set VLAN package. NOTICE: 1)Startup VLAN, if set Voice/Data VLAN differentiated as Undifferentiated, all packets will use the Voice VLAN ID as the tag. 2) Startup VLAN, if set Voice/Data VLAN differentiated as tag differentiated and disables the DiffServ, then system will not distinguish the voice and data, all packets will use the Voice VLAN ID as the tag. 3) Startup VLAN, if set Voice/Data VLAN differentiated as tag differentiated and enables the DiffServ, then system will distinguish the voice and data and add the VLAN ID each other. 4) Startup VLAN, if set Voice/Data VLAN differentiated as data untagged, then the packet of the signal/voice will use the Voice VLAN ID as the tag, but the data packets will not take the VLAN tag. 5) If Disable the VLAN, regardless to set the Voice/Data VLAN differentiated or not, all packets will not take the VLAN tag; If enable the DiffServ, all packets will only take the DiffServ value. 6) One must to notice, enable the VLAN ID Check Enable that is default, If enable it, the gateway will match the VLAN ID strictly. When others' VLAN ID not matches with us, the packets will discard. Contrarily, the gateway will accept the packets with the distinct VLAN ID. 7) You must gain the IP with the Static mode when you set VLAN, otherwise can't gain the IP in the VLAN and also can not dial with point to point. 5.3.2.4. Service Port You can set the port of telnet/HTTP/RTP by this page. SERVICE PORT Field name HTTP Port Telnet Port RTP Initial Port RTP Port Quantity explanation set web browse port, the default is 80 port,if you want to enhance system safety,you'd better change it into non-80 standard port; Example: The IP address is 192.168.1.70. and the port value is 8090, the accessing address is http://192.168.1.70:8090 Set Telnet Port, the default is 23. You can change the value into others. Example: The IP address is 192.168.1.70. the telnet port value is 8023, the accessing address is telnet 192.168.1.70 8023 Set the RTP Initial Port. It is dynamic allocation. Set the maximum quantity of RTP Port, the default is 200. Notice: 1)You need save the configuration and reboot the gateway after set this page. 2)If you modify the port of Telnet and HTTP, you would better set the value more than 1024 because the port value less than 1024 is system port reserved. 3)if you set 0 for the HTTP port, it will disable HTTP service. 5.3.2.5. DHCP SERVER DHCP SERVER Field name DHCP Leased Table explanation IP-MAC mapping table. If the LAN port of the gateway connects to a device, this table will show the IP and MAC address of this device. Shows the DHCP Lease Table, the unit of Lease time is Minute. Lease Table Name Start IP End IP Netmask Gateway Lease Time DNS Specify the name of the lease table Set the start IP address of the lease table Set the end IP address of the lease table, the network device connected to LAN port will get IP address between Start IP and End IP by DHCP. Set the Netmask of the lease table Set the Gateway of the lease table Set the Lease Time of the lease table Set the default DNS server IP of the lease table; Click the Add button to submit and add this lease table Select name of lease table, click the Delete button will delete the selected lease table from DHCP lease table. Select DNS Relay, the default is enabled. Click the Apply button to become effective. DNS Relay Notice: 1)The size of lease table can not be larger than the quantity of C network IP address. We recommend you to use the default lease table and not modify it. 2)If you modifies the DHCP lease table, you need save the configuration and reboot. 5.3.2.6. NTP Setting time zone and SNTP (Simple Network Time Protocol) server according to your location, you can also manually adjust date and time in this web page. SNTP Field name Server Time Zone Time Out NTP explanation Set SNTP Server IP address. Select the Time zone according to your location. Set the time out, the default is 60 seconds. Select the NTP, and click Apply to make the SNTP Times effective. 5.3.3. VOIP 5.3.3.1. SIP Config Set your SIP server in the following interface. SIP Config Field name explanation Choose line to set info about SIP, there are 3 lines to choose. You can switch by 【Load】 button. Register Status Server Name Server Address Server Port Account Name Password Phone Number Display Name Shows if the gateway has been registered the SIP server or not; or so, show Unapplied; Set the server name. Input your SIP server address. Set your SIP server port. Input your SIP register account name. Input your SIP register password. Input the phone number assigned by your VoIP service provider. Phone will not register if there is no phone number configured. Set the display name. Set proxy server IP address(Usually, Register SIP Proxy Server Address Proxy Server Port Proxy Username Proxy Password Domain Realm Enable Register Register Expire Time NAT Keep Alive Interval User Agent Signal Key Media Key Local port Ring type Hot line Number Conference Number Transfer Expire Time Enable subscribe Enable Keep Authentication NAT Keep Alive Server configuration is the same as Proxy SIP Server. But if your VoIP service provider give different configurations between Register SIP Server and Proxy SIP Server, you need make different settings.) Set your Proxy SIP server port. Input your Proxy SIP server account. Input your Proxy SIP server password. Set the sip domain if needed, otherwise this VoIP gateway will use the Register server address as sip domain automatically. (Usually it is same with registered server and proxy server IP address). Start to register or not by selecting it or not. Set expire time of SIP server register, default is 60 seconds. If the register time of the server requested is longer or shorter than the expire time set, the gateway will change automatically the time into the time recommended by the server, and register again. Set examining interval of the server, default is 60 seconds Set the user agent if have, the default is VoIP Phone 1.0 Set the key for signal encryption Set the key for RTP encryption Set sip port of each line Set ring type of each line Set hot line number of each line Configure conference number in server conference. For the gateway supports the transfer of certain special features server, set interval time between sending “bye” and hanging up after the phone transfers a call. Enable the option ,the gateway will receive the notify from the server. Enable/Disable Keep Authentication System will take the last authentication field which is passed the authentication by server to the request packet. It will decrease the server’s repeat authorization work, if it is enable. Enable/Disable keeps NAT of SIP alive. If some server refuse to register with too short Enable Via report Enable PRACK Long Contact Enable URI Convert Dial Without Register Ban Anonymous Call Enable DNS SRV Forward Type Forward Phone Number Server Type DTMF Mode RFC Protocol Edition Transport Protocol RFC Privacy interval time, and has no packets sending to device in private network to keep NAT alive, user could set this function ON. It need set the keep alive interval time less than the NAT server’s. Enable/Disable system to support RFC3581. Via report is special way to realize SIP NAT. Enable or disable SIP PRACK function, suggest use the default config. Set more parameters in contact field; connection with SEM server Convert # to %23 when send the URI. Set call out by proxy without registration; Set to ban Anonymous Call; Support DNS looking up with _sip.udp mode Select call forward mode, the default is Off z Off:Close down calling forward z Busy:If the phone is busy, incoming calls will z be forwarded to the appointed phone. No answer: If there is no answer, incoming z calls will be forwarded to the appointed phone. Always:Incoming calls will be forwarded to the appoint phone directly. The phone will Prompt the incoming while doing forward. Appoint your forward phone number. Select the special type of server which is encrypted, or has some unique requirements or call flows. Select DTMF sending mode, there are three modes: z DTMF_RELAY z DTMF_RFC2833 z DTMF_SIP_INFO Different VoIP Service providers may provide different modes. Select SIP protocol version to adapt for the SIP server which uses the same version as you select. For example, if the server is CISCO5300, you need to change to RFC2543; else phone may not cancel call normally. System uses RFC3261 as default. Set transport protocols, TCP or UDP; Set Anonymous call out safely; Support Edition Subscribe Expire Time Enable Conference number MWI Number Click to Talk Signal Encode RTP Encode Enable Session Timer Answer With Single Codec Auto TCP Enable Strict Proxy Enable GRUU Enable Display name Quote RFC3323and RFC3325; Overtime of resending subscribe packet. Suggest using the default config. Set to use sever conference. Input the number of the server's voice-mail box Set click to Talk (need practical software support). Enable/Disable Signal Encrypt. Enable/Disable RTP Encrypt. Set Enable/Disable Session Timer, whether support RFC4028.It will refresh the SIP sessions. Enable/Disable the function when call is incoming, phone replies SIP message with just one codec which phone supports. Set to use automatically TCP protocol to guarantee usability of transport as message is above 1300 byte Support the special SIP server-when phone receives the packets sent from server, phone will use the source IP address, not the address in via field. Set to support GRUU Set to make quotation mark to display name as the phone sends out signal, in order to be compatible with server. 5.3.3.2. Stun Config In this web page, you can config SIP STUN. STUN: By STUN server, the gateway in private network could know the type of NAT and the NAT mapping IP and port of SIP. The gateway might register itself to SIP server with global IP and port to realize the device both calling and being called in private network. STUN Field name STUN NAT Transverse STUN Server Addr STUN Server Port STUN Effect Time Local SIP Port explanation Shows STUN NAT Transverse estimation, true means STUN can penetrate NAT, while False means not. Set your SIP STUN Server IP address Set your SIP STUN Server Port Set STUN Effective Time. If NAT server finds that a NAT mapping is idle after time out, it will release the mapping and the system need send a STUN packet to keep the mapping effective and alive. Set the SIP port. Choose line to set info about SIP, There are 3 lines to choose. You can switch by 【Load】 button. Use Stun Enable/Disable SIP STUN. Notice: SIP STUN is used to realize SIP penetration to NAT. If your phone configures STUN Server IP and Port (default is 3478), and enable SIP Stun, you can use the ordinary SIP Server to realize penetration to NAT. 5.3.3.3. DIAL PEER setting This functionality offers you more flexible dial rule, you can refer to the following content to know how to use this dial rule. When you want to dial an IP address, the entry of IP addresses is very cumbersome, but by this functionality, you can set number 156 to replace 192.168.1.119 here. When you want to dial a long distance call to Beijing, you need dial an area code 010 before local phone number, but you can also dial number 1 instead of 010 after we make a setting according to this dial rule. For example, you want to dial 01062213123, but you need dial only 162213123 to realize your long distance call after you make this setting. To save the memory and avoid abundant input of user, add the follow functions: 1、x Match any single digit that is dialed. If user makes the above configuration, after user dials 11 digit numbers started with 13, the phone will send out 0 plus the dialed numbers automatically. 2、[] Specifies a range that will match digit. It may be a range, a list of ranges separated by commas, or a list of digits. If user makes the above configuration, after user dials 11 digit numbers started with from 135 to 139, the phone will send out 0 plus the dialed numbers automatically. Use this phone you can realize dialing out via different lines without switch in web interface. DIAL PEER Field name Phone number explanation There are two types of matching conditions: one is full matching, the other is prefix matching. In the Full matching, you need input your desired phone number in this blank, and then you need dial the phone number to realize calling to what the phone number is mapped. In the prefix matching, you need input your desired prefix number and T; then dial the prefix and a phone number to realize calling to what your prefix number is mapped. The prefix number supports at most 30 digits Set Destination address. This is optional config item. Destination If you want to set peer to peer call, please input destination IP address or domain name. If you want to use this dial rule on SIP2 line, you need input 255.255.255.255 or 0.0.0.2 in it.SIP3 into 0.0.0.3 Port Set the Signal port, the default is 5060 for SIP. Alias Set alias. This is optional config item. If you don’t set Alias, it will show no alias. Note: There are four types of aliases. 1) add: xxx, it means that you need dial xxx in front of phone number, which will reduce dialing number length. 2) all: xxx, it means that xxx will replace some phone number. 3) del: It means that phone will delete the number with length appointed. 4) Rep: It means that phone will replace the number with length and number appointed. You can refer to the following examples of different alias application to know more how to use different aliases and this dial rule. Call Mode Select different signal protocol, SIP or IAX2 Suffix Set suffix, this is optional config item. It will show no suffix if you don’t set it. Delete Length Set delete length. This is optional config item. For example: if the delete length is 3, the phone will delete the first 3 digits then send out the rest digits. You can refer to examples of different alias application to know how to set delete length. Examples of different alias application Set by web explanation example You need set phone number, Destination, Alias and Delete Length. Phone number is XXXT; Destination is 255.255.255.255 (0.0.0.2) and Alias is del. This means any phone No. that starts with your set phone number will be sent via SIP2 line after the first If you dial “93333”, the SIP2 server will receive “3333” several digits of your dialed phone number are deleted according to delete length. This setting will realize speed dial function, after you dialing the numeric key “2”, the number after all will be sent out. When you dial “2”, the SIP1 server will receive 33334444 The phone will automatically send out alias number adding your dialed number, if your dialed number starts with your set phone number. When you dial “8309“, the SIP1 server will receive “07558309” You need set Phone Number, Alias and Delete Length. Phone number is XXXT and Alias is rep:xxx If your dialed phone number starts with your set phone number, the first digits same as your set phone number will be replaced by the alias number specified and New phone number will be send out. When you dial “0106228”, the SIP1 server will receive “86106228” If your dialed phone number starts with your set phone number. The phone will send out your dialed phone number adding suffix number. When you dial “147”, the SIP1 server will receive “1470011” 5.3.4. Phone 5.3.4.1. DSP Config In this page, you can configure voice codec, input/output volume and so on. DSP Configuration Field name First Codec Second Codec Third Codec Forth Codec Fifth Codec Input Volume Hands-free Volume G729 Payload Length Handdown Time Ring Type Output Volume explanation The fist preferential DSP codec: G.711A/u, G.722, G.723, G.729 The second preferential DSP codec: G.711A/u, G.722, G.723, G.729 The third preferential DSP codec: G.711A/u, G.722, G.723, G.729 The forth preferential DSP codec: G.711A/u, G.722, G.723, G.729 The fifth preferential DSP codec: G.711A/u, G.722, G.723, G.729 Specify Input (MIC) Volume grade.; Specify Hands-free Volume grade Set G729 Payload Length Specify the least reflection time of Handdown, the default is 200ms. Select Ring Type Specify Output (receiver) Volume grade. Ring Volume G722 Timestamps G723 Bit Rate Default Ring Type Signal Standard VAD Specify Ring Volume grade 160/20ms or 320/20ms is available 5.3kb/s or 6.3kb/s is available Set up the ring by default Select Signal Standard. Select it or not to enable or disable VAD. If enable VAD, G729 Payload length could not be set over 20ms. 5.3.4.2. Call Service In this web page, you can configure Hotline, Call Transfer, Call Waiting, 3 Ways Call, Black List, white list Limit List and so on. Call Service Field name Hotline Warm line time No Answer Time P2P IP Prefix Do Not Disturb Ban Outgoing Enable Call explanation Specify Hotline number. If you set the number, you can not dial any other numbers. Automatically after configuration hooks time to call the hotline number. If configured to 0, the hook immediately after the call the hotline number Specify No Answer Time Set Prefix in peer to peer IP call. For example: what you want to dial is 192.168.1.119, If you define P2P IP Prefix as 192.168.1., you dial only #119 to reach 192.168.1.119. Default is “.”. If there is no “.” Set, it means to disable dialing IP. Select NO Disturb, the phone will reject any incoming call, the callers will be reminded by busy, but any outgoing call from the phone will work well. If you select Ban Outgoing to enable it, and you can not dial out any number. Enable Call Transfer by selecting it. Transfer Enable Call Waiting Enable Three Way Call Accept Any Call Black List Enable Call Waiting by selecting it. Enable Three Way Call If select it, the phone will accept the call even if the called number is not belong to the phone. Set Add/Delete Black list. If user does not want to answer some phone calls, add these phone numbers to the Black List, and these calls will be rejected. X and. Are wildcard. x means matching any single digit. for example, 4xxx expresses any number with prefix 4 which length is 4 will be forbidden to dialed out DOT (.) means matching any arbitrary number digit. for example, 6. Expresses any number with prefix 6 will be forbidden to dial out. If user wants to allow a number or a series of number incoming, he may add the number(s) to the list as the white list rule. the configuration rule is -number, for example, -123456, or -1234xx Means any incoming number is forbidden except for 4119 Note: End with DOT (.) when set up the white list Set Add/Delete Limit List. Please input the prefix of those phone Limit List numbers which you forbid the phone to dial out. For example, if you want to forbid those phones of 001 as prefix to be dialed out, you need input 001 in the blank of limit list, and then you can not dial out any phone number whose prefix is 001. X and. Are wildcard. X means matching any single digit. for example, 4xxx expresses any number with prefix 4 which length is 4 will be forbidden to dialed out . Means matching any arbitrary number digit. For example, 6. Expresses any number with prefix 6 will be forbidden to dial out. Notice: Black List and Limit List can record at most10 items respectively. 5.3.4.3. Digital Map Configuration This system supports 4 dial modes: 1). End with “#”: dial your desired number, and then press #. 2). Fixed Length: the phone will intersect the number according to your specified length. 3). Time Out: After you stop dialing and waiting time out, system will send the number collected. 4). User defined: you can customize digital map rules to make dialing more flexible. It is realized by defining the prefix of phone number and number length of dialing. In order to keep some users' secondary dialing manner when dialing the external line with PBX, phone can be added a special rule to realize it. So user can dial a number as external line prefix and get the secondary dial tone to keep dial the external number. After finishing dialing, phone will send the prefix and external number totally to the server. For example, there is a rule 9, xxxxxxxx in the digital map table. After dialing 9, phone will send the secondary dial tone, user may keep going dialing. After finished, phone will call the number which starts with 9; actually the number sent out is 9-digit with 9. Digital Map Configuration Field name End with "#" Fixed Length Time out explanation Set Enable/Disable the phone ended with “#” dial. Specify the Fixed Length of phone ending with. Set the timeout of the last dial digit. The call will be sent after timeout. Below is user-defined digital map rule: [] Specifies a range that will match digit. May be a range, a list of ranges separated by commas, or a list of digits. x Match any single digit that is dialed. . Match any arbitrary number of digits including none. Tn Indicates an additional time out period before digits are sent of n seconds in length. n is mandatory and can have a value of 0 to 9 seconds. Tn must be the last 2 characters of a dial plan. If Tn is not specified it is assumed to be T0 by default on all dial plans. Cause extensions 1000-8999 to be dialed immediately Cause 8 digit numbers started with 9 to be dialed immediately Cause 911 to be dialed immediately after it is entered. Cause 99 to be dialed after 4 seconds. Cause any number started with 9911 to be dialed 4 seconds after dialing ceases. Notice: End with “#”, Fixed Length, Time out and Digital Map Table can be used simultaneously, System will stop dialing and send number according to your set rules. 5.3.5. Maintenance 5.3.5.1. Auto Provision Auto Provision Field name Current Config Version Server Address Username Password Config File Name Config Encrypt explanation Show the current config file’s version. Set FTP/TFTP/HTTP server IP address for auto update. The address can be IP address or Domain name with subdirectory. Set FTP server Username. System will use anonymous if username keep blank. Set FTP server Password. Set configuration file’s name which need to update. System will use MAC as config file name if config file name keep blank. For example, 000102030405.。 Input the Encrypt Key, if the configuration file is Key Protocol Type Update Interval Time Update Mode encrypted. Select the Protocol type FTP、TFTP or HTTP. Set update interval time, unit is hour. Different update modes: 1. Disable: means no update 2. Update after reboot: means update after reboot. 3. Update at time interval: means periodic update. 5.3.5.2. Syslog Config Syslog is a protocol which is used to record the log messages with client/server mechanism. Syslog server receives the messages from clients, and classifies them based on priority and type. Then these messages will be written into log by some rules which administrator can configure. This is a better way for log management. 8 levels in debug information: Level 0---emergency: This is highest default debug info level. You system can not work. Level 1---alert: Your system has deadly problem. Level 2---critical: Your system has serious problem. Level 3---error: The error will affect your system working. Level 4---warning: There are some potential dangers. But your system can work. Level 5---notice: Your system works well in special condition, but you need to check its working environment and parameter. Level 6---info: the daily debugging info. Level 7---debug: the lowest debug info. Professional debugging info from R&D person. At present, the lowest level of debug information send to Syslog is info; debug level only can be displayed on telnet. Syslog Configuration Field name Server IP Server Port MGR Log Level explanation Set Syslog server IP address. Set Syslog server port. Set the level of MGR log. SIP Log Level IAX2 Log Level Enable Syslog Set the level of SIP log. Set the level of IAX2 log. Select it or not to enable or disable syslog. 5.3.5.3. Config Setting Config Setting Field name Save Config Backup Config Clear Config explanation you can save all changes of configurations. Click the Save button, all changes of configuration will be saved, and be effective immediately. . Right clicks on “Right click here…” and select “Save Target As….” then you will save the config file in .txt format user can restore factory default configuration and reboot the gateway. If you login as Admin, the gateway will reset all configurations and restore factory default; if you login as Guest, the gateway will reset all configurations except for VoIP accounts (SIP1-2 and IAX2) and version number. 5.3.5.4. Update You can update your configuration with your config file in this web page. Update Field name explanation Click the browse button, find out the config file saved Web Update before or provided by manufacturer, download it to the gateway directly, press “Update” to save. You can also update downloaded update file, logo picture, ring, mmiset file by web. Server Set the FTP/TFTP server address for download/upload. The address can be IP address or Domain name with subdirectory. Username Set the FTP server Username for download/upload. Password Set the FTP server password for download/upload. File name Set the name of update file or config file. The default name is the MAC of the gateway, such as 000102030405. Notice: You can modify the exported config file. And you can also download config file which includes several modules that need to be imported. For example, you can download a config file just keep with SIP module. After reboot, other modules of system still use previous setting and are not lost. Action type that system want to execute: Type Protocol 1. Application update: download system update file 2. Config file export: Upload the config file to FTP/TFTP server, name and save it. 3. Config fie import: Download the config file to gateway from FTP/TFTP server. The configuration will be effective after the gateway is reset. Select FTP/TFTP server 5.3.5.5. Account Config You can add or delete user account, and change the authority of each user account in this web page Account Configuration Field name Keyboard Password explanation Set the password for entering the setting menu of the gateway by the phone key board. The password is digit. This table shows the current user existed. User Name Set account user name. User Level Set user level, Root user has the right to modify configuration, General can only read. Password Set the password. Confirm Confirm the password. Select the account and click the Modify to modify the selected account, and click the Delete to delete the selected account. General user only can add the user whose level is General. 5.3.5.6. Reboot If you modified some configurations which need the gateway’s reboot to be effective, you need click the Reboot, then the gateway will reboot immediately. Notice: Before reboot, you need confirm that you have saved all configurations.. 5.3.6. Security 5.3.6.1. MMI Filter MMI Filter User could make some device own IP, which is pre-specified, access to the MMI of the gateway to config and manage the gateway. Field name explanation MMI Filter IP Table list: Add or delete the IP address segments that access to the phone. Set initial IP address in the Start IP column, Set end IP address in the End IP column, and click Add to add this IP segment. You can also click Delete to delete the selected IP segment. MMI Filter Select it or not to enable or disable MMI Filter. Click Apply to make it effective. Notice: Do not set your visiting IP outside the MMI filter range, otherwise, you can not logon through the web. 5.3.6.2. Firewall Firewall Configuration In this web interface, you can set up firewall to prevent unauthorized Internet users from accessing private networks connected to the Internet (input rule), or prevent unauthorized private network devices from accessing the Internet (output rule). Firewall supports two types of rules: input access rule and output access rule. Each type supports at most 10 items. Through this web page, you could set up and enable/disable firewall with input/output rules. System could prevent unauthorized access, or access other networks set in rules for security. Firewall, is also called access list, is a simple implementation of a Cisco-like access list (firewall). It supports two access lists: one for filtering input packets, and the other for filtering output packets. Each kind of list could be added 10 items. We will give you an instance for your reference. Field name In access enable out access enable Input/Output Deny/Permit explanation Select it to Enable in_ access rule Select it to Enable out_ access rule Specify current adding rule by selecting input rule or output rule. Specify current adding rule by selecting Deny rule or Permit rule. Protocol Type Filter protocol type. You can select TCP, UDP, ICMP, or IP. Port Range Src Addr Set the filter Port range Set source address. It can be single IP address, network address, complete address 0.0.0.0, or network address similar to *.*.*.0 Des Addr Set the destination address. It can be IP address, network address, complete address 0.0.0.0, or network address similar to *.*.*.* the source address’ mask. For example, 255.255.255.255 means just point to one host; 255.255.255.0 means point to a network which network ID is C type. Set the destination address’ mask. For example, Des Mask 255.255.255.255 means just point to one host; 255.255.255.0 means point to a network which network ID is C type. Click the Add button if you want to add a new output rule. Set Src Mask Then enable out access, and click the Apply button. So when devices execute to ping 192.168.1.118, system will deny the request to send ICMP request to 192.168.1.118 for the out access rule. But if devices ping other devices which network ID is 192.168.1.0, it will be normal. Click the Delete button to delete the selected rule. 5.3.6.3. NAT Config NAT is abbreviated from Net Address Translation; it’s a protocol responsible for IP address translation. In other word, it is responsible for transforming IP and port of private network to public, also is the IP address mapping which we usually say. NAT Configuration Field name IPSec ALG FTP ALG PPTP ALG explanation It is an encryption technology. Select it to enable IPSec ALG, the default is enable FTP is a service of connection layer which can transform intranet IP into extranet IP when intranet IP is sending out packet. Select it to enable FTP ALG, the default is enable Select it enable PPTP ALG, the default is enable Shows the NAT TCP mapping table Shows the NAT UDP mapping table Transfer Type Inside IP Select the NAT mapping protocol style, TCP or UDP Set the IP address of device which is connected to LAN interface to do NAT mapping. Inside Port Set the LAN port of the NAT mapping Outside Port Set the WAN port of the NAT mapping Notice: After finish setting, click the Add button to add new mapping table; click the Delete button to delete the selected mapping table. 5.3.6.4. VPN Config This web page provides us a safe connect mode by which we can make remote access to enterprise inner network from public network. That is to say, you can set it to connect public networks in different areas into inner network via a special tunnel. VPN Configuration Field name VPN IP explanation Shows the current VPN IP address Select UDP Tunnel (VPN Tunnel) or VPN L2TP. You can choose only one for current state. After you select it, you’d better save configuration and reboot your device. Enable VPN Select it or not to enable or disable VPN; VPN Server Addr VPN User Name VPN Password Set VPN L2TP Server IP address Set User Name access to VPN L2TP Server Set Password access to VPN L2TP Server VPN Server Addr VPN User Name VPN Password Set VPN PPTP Server IP address Set User Name access to VPN PPTP Server Set Password access to VPN PPTP Server 5.3.7. Logout Click Logout,and you will exit web page. If you want to enter it next time, you need input user name and password again. 6. Appendix 6.1. Specification 6.1.1. Hardware Item Adapter (Input/Output) port WAN LAN Operation Temperature Relative Humidity main chip SDRAM Flash A1 GATEWAY Input: 100-240V Output: 12V 1A 10/100Base- T RJ-45 for LAN 10/100Base- T RJ-45 for PC 0~40℃ 10~65% Ralink MIPS 24KEC (320MHz) 16M 4M 6.1.2. Voice features z z z z z z z z z z z z z z z z Support SIP 2.0 (RFC3261) and correlative RFCs Codec: G.711A/u, G.729a/b,G.726-32k,ilbc Echo cancellation: G.168 Compliance in LEC, additional acoustic echo cancellation(AEC) can reach 96ms max filter length in hands-free mode Support Voice Gain Setting, VAD, CNG NAT penetration, Support for STUN way through SIP support SIP domain, SIP authentication(none basic, MD5), DNS name of server, Peer to Peer/ IP call SIP can register two SIP accounts, through the Pubic Server / Private server, users can either account for inbound and outbound Support call line automatically selected, when the public can not connect the server when the server can automatically switch to the private call DTMF Relay: support SIP info,DTMF Relay,RFC2833 SIP application: SIP Call forward/transfer(blind/attended)/hold/waiting/3 way talking/ Call control features: Flexible dial map, hotline, empty calling No. reject service, black list for reject authenticated call, limit call, no disturb, caller ID, Flexible deer peer rule. Support T.38 Fax With the escape port (lifeline), can support power to answer and make phone calls through the exit port can also be the system starts the call by dialing rules lifeline Add voip unavailable features to automatically connect to the lifeline routes Add busy when N / A lines of the 4 modes Support IAX2 6.1.3. Network features z z WAN/LAN: support bridge and router model Support PPPoE for XDSL z z z z z z z z z z z Support DHCP server in the LAN port Gateway ping test through keyboard commands Support DHCP client in the WAN port Support basic NAT and NAPT Support VLAN (optional: voice vlan/ data vlan),support NTP Support DMZ Support VPN (L2TP) function WAN Port supports main DNS and secondary DNS server can select dynamically to get DNS in DHCP mode or statically set DNS address. QoS with DiffServ Support DNS relay, supports SNTP Client, Firewall support the simple Network tools in telnet server: including ping, trace route, telnet client 6.1.4. Maintenance and management z z z z z z z Support Safe Mode Can be updated by safe Mode Web ,telnet and keypad management Management with different account right Upgrade firmware through HTTP, FTP or TFTP Telnet remote management/ upload/download setting file Support Syslog Support Auto Provisioning (upgrade firmware or configuration file) 6.2. Particularly suitable for A1 single port gateway z z z z z z z z z Service Provider of telecom operators and (ITSP) Internet Telephony Large companies (for international and domestic long distance and / or internal communications, mainly in the way free sparring) Import and export business of small or medium enterprises, such as foreign travel, study intermediary agents, immigration agents and other intermediaries Foreign / joint ventures, foreign enterprises in China, offices, representatives and agents, etc. Foreign hotel (which can be placed in the rooms and business center or leased) All levels of government in dealing with foreigners more departments, such as foreign trade sector, the CPAFFC, sports units, cultural units, Foreign Experts Affairs, the foreign affairs department, etc. Schools and research institutes, such as the joint venture school, school or Foreign Affairs Department of the research unit. IP supermarkets, IP telephone booth (mostly set in the migrant workers, students focus on areas such as low-income people) Personal and home users, such as immigrant families, host families, student hostels, separation of individual family members due to long working relationship, often with family or friends living abroad keep in touch with the individuals. 6.3. Common Problems Symptom POWER light does not shine Solution 1、Check the power connection is correct. 2、Check the power adapter is used. 1、Check the cable connection is valid, check the PC card indicator light is on. 49 WAN/LAN link light does not shine Can not access the internet 2 、 Check the card is working properly, the specific approach is seen in the PC, there with "?" Or "!" Device under "Network Adapter". If so, remove the device and reinstall. Otherwise, the NIC in another slot, if not enough, replace the card. Access modes commonly used example (already installed on your computer dial-up software) Description: 1、Make sure the front of the problem does not exist. 2、Make sure that dial-up software is properly installed and set. 3、Sure to enter the correct user name and password. 4、If it does not work after the success dial up, make sure the IE browser's proxy server is set correctly. 5、Please try to log multiple pages to confirm a Web server failure is not due. 50